Voice over internet protocol system having dynamic gain control function and method thereof

CLAIM OF PRIORITY

p-0002This application makes reference to, and claims all benefits accruing under 35 U.S.C. §119 from an application for VOICE OVER INTERNET PROTOCOL SYSTEM ENABLE TO CONTROL GAIN DYNAMICALLY AND METHOD THEREOF earlier filed in the Korean Intellectual Property Office on 17 Feb. 2003 and there duly assigned Serial No. 2003-9912.

BACKGROUND OF THE INVENTION

p-00031. Field of the Invention

p-0004The present invention relates to a Voice over Internet Protocol system and, more particularly, to a Voice over Internet Protocol system having a dynamic gain control function and a method for providing a dynamic gain using the system wherein in the process of converting Pulse Code Modulation (PCM) data to a Voice over Internet Protocol packet or vice versa, a dynamic gain value is assigned in accordance with the type of call, and PCM data can be amplified and outputted in accordance with the assigned gain value.

p-00052. Description of the Related Art

p-0006Generally, communication apparatuses connected to telephone lines include a general telephone, a facsimile, and so on. A signal intensity of a communication apparatus connected to a telephone line is changed in accordance with the distance from an exchange. That is, since the telephone line made up of a general copper wire has impedance values changing depending upon its length, a loop current relative to a fixed direct current (DC) of the exchange also changes.

p-0007Since it may be said that the change of the loop current indicates the change of the length, the change of the signal according to the distance from the exchange is proportional to the change of the loop current. So, a communication equipment user separated far from the exchange hears the other party's voice as a very low sound or never hears it under certain circumstances.

p-0008Accordingly, it is necessary to compensate a signal on the attenuating telephone line according to its distance from the exchange. And, it is possible to have the other party's voice heard well even in communication equipment separated far from the exchange by preparing a gain control circuit in the exchange and controlling a gain of a signal received from a terminal.

p-0009<figref idrefs="DRAWINGS">FIG. 1</figref> is a view showing a representative block construction of an exemplary subscriber terminal gain control circuit of an exchange in the art.

p-0010Referring to <figref idrefs="DRAWINGS">FIG. 1</figref>, the gain control circuit of the exchange subscriber terminal includes a subscriber connector <b>110</b> for connecting the exchange to a subscriber terminal, and an amplification part <b>120</b> for amplifying a signal provided from the subscriber connector <b>110</b> to a desired level.

p-0011A subscriber terminal (not shown) to be subscribed and connected to the exchange is connected to the exchange through the subscriber connector <b>110</b>.

p-0012A voice signal applied from the terminal through the subscriber connector <b>110</b> is applied to and amplified in the amplification part <b>120</b>. Here, an amplifier <b>126</b> decides an amplification gain by a ratio of a first resistor <b>122</b>, an input resistor, and a second resistor <b>124</b>, an output resistor, and the gain can be calculated according to the following expression. <br />amplified gain=amplification gain ratio*20 Log (<i>R</i>2/<i>R</i>1)  (Expression 1)

p-0013Here, R<b>1</b> and R<b>2</b> indicate an input resistor and an output resistor respectively. As shown in expression 1, the amplification gain can be variable by properly controlling the ratio of the first resistor <b>122</b> of the input resistor and the second resistor <b>124</b> of the output resistor.

p-0014An analog voice signal amplified and applied from the amplification part <b>120</b> is applied to PCM converter <b>128</b> to be converted to a digital signal, and the converted digital signal is applied to a controller <b>130</b> so as to perform a process needed for exchanging.

p-0015However, in case of using the resistor <b>122</b> and the resistor <b>124</b> which have fixed values respectively, receiver sensitivity is transmitted as it is, regardless of the condition of a cable since fixed transmission and receiver gains are maintained despite the change of input state due to an external condition.

p-0016Since a call gain control circuit in the art has only a unidirectional gain value using fixed elements, it becomes difficult to control the transmission and receiver gains in accordance with characteristic of each line in the real situation that may have various states.

p-0017In order to solve this problem, a programmable gain control circuit has been developed wherein transmission and reception lines each have a number of resistors which are the same for each line in number and serially connected to each line and on/off switches are connected in parallel to the resistors so that the on/off of the switches are controlled by the controller in accordance with states of the cables and then the values of the resistors to determine the controlled transmission and receiver gains.

p-0018On the other hand, a Voice over Internet Protocol (referred to as a VoIP, hereinafter) is a communication service of new mode wherein a voice communication is performed not through an existing communication network, a Public Switch Telephone Network (referred to as a PSTN, hereinafter), but through an Internet network. Since the communication method using the Internet network uses a packet-based network different from existing communication methods, a user does not have to pay for charges of domestic/international phone lines separately so that it is possible to perform the voice communication at a lower fare.

p-0019The VoIP has a faculty of transmitting video information as well as audio information using an H.323 Protocol being an ITU-T (International Telecommunication Union—Telecommunication) standard that provides fundamental principles for voice, video, and data communications over the IP (Internet Protocol) network including the Internet. One of the H.323 entities defined in the H.323 protocol is a gatekeeper. The gatekeeper binds H.323 endpoints present in a packet-based network (i.e., an IP-based network) in one control zone defined as a “Zone”, and then controls/manages the bound H.323 endpoints.

p-0020A VoIP system using the Internet network as a back-bone has an exemplary construction as shown in <figref idrefs="DRAWINGS">FIG. 2</figref>.

p-0021Referring to <figref idrefs="DRAWINGS">FIG. 2</figref>, the VoIP system has an Internet <b>208</b> used as the back-bone, and the Internet <b>208</b> is connected to gateways <b>206</b> and <b>210</b> and to personal computers <b>216</b> and <b>218</b> (referred to as PCs, hereinafter).

p-0022The gateways <b>206</b> and <b>210</b> are correspondingly connected to PSTNs <b>204</b> and <b>212</b> that also are connected to telephone terminals <b>202</b> and <b>214</b> respectively. Terminals such as the phone terminals <b>202</b> and <b>214</b> and the PCs <b>216</b> and <b>218</b> are endpoints which are capable of communicating by voices (essential), images (option) and data (option) during a one-to-one communication or a conference.

p-0023Such terminals can perform a real-time and a bidirectional communication with the gateways <b>206</b> and <b>210</b> and other terminals. The gateways <b>206</b> and <b>210</b> are elements that enable terminals (for example, PCs <b>216</b> and <b>218</b>) connected to the Internet <b>208</b> being a packet-based network and terminals (for example, telephone terminals <b>202</b> and <b>214</b>) connected to the PSTNs <b>204</b> and <b>212</b> or an Integrated Service Digital Network (referred to as an ISDN, hereinafter) to perform the real-time and the bidirectional communication.

p-0024Briefly, gateways <b>206</b> and <b>210</b> perform a real-time compression and a protocol transformation of voices and facsimile data inputted from the PSTNs <b>204</b> and <b>212</b> and send the data to the Internet <b>208</b>.

p-0025The Internet phone (IP-phone) can be classified into 3 types according to the kind of terminal used at both ends, that is, PC to PC, PC to phone and phone to phone.

p-0026Generally, the technical principle of the Internet phone is made up of a voice encoding and compressing technology, a real-time data transmission technology, a packet recovery technology, a gateway technology, and so on.

p-0027The voice encoding technology employs a low bit rate, high compression rate and high voice quality encoding technique in order to transmit the voice information without damaging the voice quality.

p-0028The voice encoding technology includes a PCM, an adaptive prediction coding, a Global System for Mobile communication (referred to as GSM, hereinafter), a Linear Predictive Coding (referred to as LPC, hereinafter), and so on, and the above technologies are now used. The real-time transformation technology includes a Real Time Transport Protocol (referred to as RTP, hereinafter). The RTP receives much recognition in the transmission quality over the Internet and is mainly used since 1995.

p-0029Also, a gateway embodying technology is to embody a gateway which is a network connection apparatus transforming analog voice information in order to transmit the information from an existing line exchange network to a packet exchange network.

p-0030A basic function of the gateway is processed in a digital signal processor. The gateway performs a voice compression capability using a compression algorithm, a waiting capability, and a removal capability so that it is possible to transform and transmit PCM voice data to a VoIP packet, and vice versa.

p-0031Here, in the process that the digital signal processor transforms the PCM data into the VoIP packet by performing the voice compression capability using the compression algorithm, the waiting capability, and the removal capability, the strength of voice may become too low or too high. Therefore, the gateway has a capability of controlling a gain value before compressing the PCM data to the VoIP packet, like the gain controlling circuit in the exchange described above.

p-0032However, since the gain value in the gateway is already assigned by the operator as a fixed value (of course, the gain value is generally determined by a test according to an environmental characteristic of the gateway), it is difficult to dynamically change the gain value according to the kind of call.

SUMMARY OF THE INVENTION

p-0033Therefore, the present invention has been made in view of the above problem, and it is an object of the present invention to provide a VoIP system having a dynamic gain control capability and a method for providing a dynamic gain using the system wherein in case that the VoIP gateway operates in an interworking state with a key telephone or a private exchange, a gain value can be dynamically assigned according to the type of call.

p-0034In accordance with an aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a private exchange, on receiving a call setup signal from an extension telephone, for determining the type of telephone using an extension number of the telephone and transmitting a gain value assigned according to the type of determined telephone; and a VoIP gateway for a) receiving the gain value from the private exchange, storing the value, amplifying PCM voice data transmitted from the private exchange according to the stored gain value when a call setup is completed, transforming the amplified PCM voice data into a VoIP packet and transmitting the packet to a receiver, and b) transforming the VoIP packet transmitted from the receiver into PCM voice data, amplifying the transformed PCM voice data according to the stored gain value, and transmitting the amplified data to the private exchange.

p-0035In accordance with another aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a private exchange, on receiving a call setup signal from an extension telephone, for determining the type of receiver using a telephone number of the receiver and transmitting a gain value assigned according to the type of determined receiver; and a VoIP gateway for a) receiving the gain value from the private exchange, storing the value, amplifying PCM voice data transmitted from the private exchange according to the gain value stored when a call setup is completed, transforming the amplified PCM voice data into a VoIP packet and transmitting the packet to the receiver, and b) transforming the VoIP packet transmitted from the receiver into PCM voice data, amplifying the transformed PCM voice data according to the stored gain value, and transmitting the amplified data to the private exchange.

p-0036In accordance with another aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a private exchange, on receiving a call setup signal from an extension telephone, for determining the type of telephone using an extension number, determining the type of receiver using a telephone number of the receiver, and transmitting a gain value assigned according to the type of determined receiver and the receiver; and a VoIP gateway for a) receiving the gain value from the private exchange, storing the value, amplifying PCM voice data transmitted from the private exchange according to the stored gain value when a call setup is completed, transforming the amplified PCM voice data into a VoIP packet and transmitting the packet to the receiver, and b) transforming the VoIP packet transmitted from the receiver into the PCM voice data, amplifying the transformed PCM voice data according to the stored gain value, and transmitting the amplified data to the private exchange.

p-0037Preferably, the system in accordance with the present invention further comprises a gatekeeper for, on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of telephone using the telephone number of the receiver and transmitting the gain value assigned according to the type of determined telephone to the VoIP gateway, and wherein the VoIP gateway a) stores the gain value transmitted from the gatekeeper, transforms the VoIP packet transmitted from the sender to the PCM voice data when the call setup is completed, amplifies the transformed PCM voice data and transmits the data to the private exchange, and b) amplifies the PCM voice data transmitted from the private exchange according to the stored gain value, transforms the amplified data into the VoIP packet and transmits the data to the sender.

p-0038Preferably, the system in accordance with the present invention further comprises a gatekeeper for, on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of sender using an IP address of the sender, and transmitting the gain value assigned according to the determined sender to the VoIP gateway, and wherein the VoIP gateway a) stores the gain value transmitted from the gatekeeper, transforms the VoIP packet <b>11</b> transmitted from the sender to the PCM voice data when the call setup is completed, amplifies the transformed PCM voice data and transmits the data to the private exchange, and b) amplifies the PCM voice data transmitted from the private exchange according to the stored gain value, transforms the amplified data into the VoIP packet and transmits the data to the sender.

p-0039Preferably, the system in accordance with the present invention further comprises a gatekeeper for, on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of telephone of the receiver using the telephone number of the receiver, determining the type of sender using the IP address of the sender, and transmitting the gain value assigned according to the type of telephone of the determined sender and receiver to the VoIP gateway, and wherein the VoIP gateway a) stores the gain value transmitted from the gatekeeper, transforms the VoIP packet transmitted from the sender to the PCM voice data when the call setup is completed, amplifies the transformed PCM voice data and transmits the data to the private exchange, and b) amplifies the PCM voice data transmitted from the private exchange according to the stored gain value, transforms the amplified data into the VoIP packet and transmits the data to the sender.

p-0040Preferably, in the system in accordance with the present invention, the VoIP gateway includes: a Public Switched Telephone Network (PSTN) connector for providing an interface with the private exchange; an Internet network connector for providing a connection with an Internet network; a media processor for a) amplifying the PCM voice data transmitted from the private exchange through the PSTN connector according to the established gain value when the call setup is completed, transforming the amplified data into the VoIP packet and transmitting the packet to the receiver through the Internet network connector, and b) transforming the VoIP packet transmitted from the receiver through the Internet network connector into the PCM voice data, amplifying the transformed PCM voice data according to the gain value, and transmitting the amplified data to the private exchange through the PSTN connector; and a main controller for receiving the gain value from the private exchange through the PSTN connector, storing the gain value, providing the media processor with the gain value when the call setup is completed, and amplifying the PCM voice data according to the stored gain value.

p-0041Preferably, in the system in accordance with the present invention, the media processor includes: a memory for storing the gain value transmitted from the main controller; a digital signal processor for a) amplifying the PCM voice data transmitted from the private exchange through the PSTN connector according to the gain value when the call setup is completed, transforming the amplified data into the VoIP packet and transmitting the packet to the receiver through the Internet network connector, and b) transforming the VoIP packet transmitted from the receiver through the Internet network connector into the PCM voice data, amplifying the transformed PCM voice data according to the gain value and transmitting the data to the PSTN connector; and a controller for receiving the gain value from the main controller, storing the gain value in the memory, providing the digital signal processor with the gain value when the call setup is completed, and amplifying the PCM voice data according to the gain value.

p-0042Preferably, in the course that the private exchange transmits the gain value to the VoIP gateway, the private exchange transmits the gain value using a call setup message.

p-0043In accordance with another aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a gatekeeper for, on receiving a call setup signal whose receiver is an extension telephone of a private exchange, determining the type of sender using a telephone number of the sender and transmitting a gain value assigned according to the type of determined sender; and a VoIP gateway for a) storing the gain value transmitted from the gatekeeper, transforming a VoIP packet transmitted from the sender into PCM voice data when a call setup is completed, and amplifying the transformed PCM voice data and transmitting the data to the private exchange, and b) amplifying the PCM voice data transmitted from the private exchange according to the stored gain value, transforming the data into the VoIP packet and transmitting the packet to the sender.

p-0044In accordance with another aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a gatekeeper for, on receiving a call setup signal whose receiver is an extension telephone of a private exchange, determining the type of sender using an IP address of the sender and transmitting a gain value assigned according to the determined sender; and a VoIP gateway for a) storing the gain value transmitted from the gatekeeper, transforming a VoIP packet transmitted from the sender into PCM voice data when a call setup is completed, amplifying the transformed PCM voice data and transmitting the data to the private exchange, and b) amplifying the PCM voice data transmitted from the private exchange according to the stored gain value, transforming the data into the VoIP packet, and transmitting the packet to the sender.

p-0045In accordance with another aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a gatekeeper for, on receiving a call setup signal whose receiver is an extension telephone of a private exchange, determining the type of telephone of a receiver using a telephone number of the receiver, determining the type of sender using an IP address of the sender, and transmitting a gain value assigned according to the type of telephone of the determined receiver and the determined sender; and a VoIP gateway for a) storing the gain value transmitted from the gatekeeper, transforming a VoIP packet transmitted from the sender into the PCM voice data when a call setup is completed, amplifying the transformed PCM voice data, and transmitting the data to the private exchange, and b) amplifying the PCM voice data transmitted from the private exchange according to the stored gain value, transforming the data into the VoIP packet, and transmitting the packet to the sender.

p-0046Preferably, in the course that the gatekeeper transmits the gain value to the VoIP gateway, the gatekeeper transmits the gain value using a call setup message.

p-0047In accordance with another aspect of the present invention, there is provided a method for providing a gain dynamically in a Voice over Internet Protocol (VoIP) system, comprising the steps of: when receiving a call setup signal from an extension telephone, determining the type of telephone by a private exchange using an extension number of the extension telephone; transmitting a gain value assigned according to the type of telephone determined in the step a) to a VoIP gateway by the private exchange; and by the VoIP gateway, storing the gain value transmitted from the private exchange and amplifying PCM voice data according to the gain value when a call setup is completed.

p-0048In accordance with another aspect of the present invention, there is provided a method for providing a gain dynamically in a Voice over Internet Protocol (VoIP) system, comprising the steps of: a) when receiving a call setup signal from an extension telephone, determining the type of receiver using a number of the receiver by a private exchange; b) transmitting a gain value assigned according to the type of receiver determined in the step a) to a VoIP gateway by the private exchange; and c) by the VoIP gateway, storing the gain value transmitted from the private exchange and amplifying PCM voice data according to the gain value when a call setup is completed.

p-0049In accordance with yet another aspect of the present invention, there is provided a method for providing a gain dynamically in a Voice over Internet Protocol (VoIP) system, comprising the steps of: a) when receiving a call setup signal from an extension telephone, by a private exchange, identifying the type of telephone of a sender using the extension number of the extension telephone and determining the type of telephone of a receiver using the number of the receiver; b) transmitting a gain value assigned according to the type of telephone of the sender and the type of telephone of the receiver determined in the step a) to a VoIP gateway by the private exchange; and c) by the VoIP gateway, storing the gain value transmitted from the private exchange and amplifying PCM voice data according to the gain value when a call setup is completed.

p-0050Preferably, the step c) includes the steps of: c-1) storing the gain value transmitted from the private exchange; c-2) amplifying the PCM voice data transmitted from the private exchange according to the gain value, transforming the data into a VoIP packet and transmitting the packet to the receiver, after the call setup is completed; and c-3) transforming the VoIP packet transmitted from the receiver into the PCM voice data, amplifying the transformed PCM voice data according to the gain value and transmitting the data to the private exchange.

p-0051Preferably, the method in accordance with the present invention further comprises the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of telephone using the telephone number of the receiver by the gatekeeper; transmitting the gain value assigned according to the determined type of telephone to the VoIP gateway by the gatekeeper; and by the VoIP gateway, storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed.

p-0052Preferably, the method in accordance with the present invention further comprises the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of sender using an IP address of the sender by the gatekeeper; transmitting the gain value assigned according to the type of sender determined in the step d) to the VoIP gateway by the gatekeeper; and by the VoIP gateway, storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed.

p-0053Preferably, the method of the present invention further comprises the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of receiver using the telephone number of the receiver and determining the type of sender using the IP address of the sender, by the gatekeeper; e) transmitting the gain value assigned according to the type of telephone of the receiver and the type of sender determined in the step d) to the VoIP gateway by the gatekeeper; and by the VoIP gateway, storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed.

p-0054Preferably, the method of the present invention further comprises the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of telephone using the telephone number of the receiver by the gatekeeper; e) transmitting the gain value assigned according to the type of telephone determined in the step d) to the VoIP gateway by the gatekeeper; and f) by the VoIP gateway, storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed.

p-0055Preferably, the method of the present invention further comprises the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of sender using the IP address of the sender by the gatekeeper; transmitting the gain value assigned according to the type of sender determined in the step d) to the VoIP gateway by the gatekeeper; and storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed by the VoIP gateway.

p-0056Preferably, the method of the present invention further comprising the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of telephone of the receiver using the telephone number of the receiver, and determining the type of sender using the IP address of the sender by the gatekeeper; transmitting the gain value assigned according to the type of telephone of the receiver and the type of sender determined in the step d) to the VoIP gateway by the gatekeeper; and storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed, by the VoIP gateway.

BRIEF DESCRIPTION OF THE DRAWINGS

p-0057A more complete appreciation of the present invention, and many of the attendant advantages thereof, will become readily apparent as the same becomes better understood by reference to the following detailed description when considered in conjunction with the accompanying drawings in which like reference symbols indicate the same or similar components, wherein:

p-0058<figref idrefs="DRAWINGS">FIG. 1</figref> is a view showing a representative block construction of an exemplary subscriber terminal gain control circuit of an exchange in the art;

p-0059<figref idrefs="DRAWINGS">FIG. 2</figref> is a view showing an exemplary construction of a VoIP system using a general Internet network as a back-bone;

p-0060<figref idrefs="DRAWINGS">FIG. 3</figref> is a view showing a VoIP system having a dynamic gain control capability in accordance with a preferred embodiment of the present invention;

p-0061<figref idrefs="DRAWINGS">FIG. 4</figref> is a view showing an inner block diagram of a VoIP gateway;

p-0062<figref idrefs="DRAWINGS">FIG. 5</figref> is a view showing a detailed block construction of an IP network connector and a media processor shown in <figref idrefs="DRAWINGS">FIG. 4</figref>;

p-0063<figref idrefs="DRAWINGS">FIG. 6</figref> is a view showing an operation flow chart of a method for providing a dynamic gain in a sender using a VoIP system in accordance with an embodiment of the present invention; and

p-0064<figref idrefs="DRAWINGS">FIG. 7</figref> is a view showing an operation flow chart of a method for providing a dynamic gain in a receiver using a VoIP system in accordance with another embodiment of the present invention.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

p-0065Now, a VoIP system having a dynamic gain control capability and a method for providing a dynamic gain using the system in accordance with a preferred embodiment of the present invention will be described in detail with reference to the annexed drawings.

p-0066<figref idrefs="DRAWINGS">FIG. 3</figref> is a view showing a VoIP system having a dynamic gain control capability in accordance with a preferred embodiment of the present invention.

p-0067Referring to <figref idrefs="DRAWINGS">FIG. 3</figref>, a VoIP system having a dynamic gain control capability in accordance with a preferred embodiment of the present invention includes a key telephone/private exchange <b>330</b>, a VoIP gateway <b>340</b>, and a gatekeeper <b>350</b>.

p-0068The key telephone/private exchange <b>330</b> is a telephone exchange system which enables a predetermined number of external telephone lines used in enterprise fields such as a public office, a company, a factory and a hotel to be shared with all members, and phone calls between internal users connected to extension lines to be connected automatically.

p-0069A main object of the key telephone/private exchange <b>330</b> is to reduce an expenditure occurring in case that all members of a public office, a company, a factory and a hotel have their own general telephone lines.

p-0070The key telephone/private exchange <b>330</b> is owned not by a telephone company but privately owned by a public office, a company, a factory and a hotel and is managed on their own responsibility. The key telephone/private exchange <b>330</b> employed an analog mode originally but the trend of it is recently changing to a digital mode.

p-0071The key telephone/private exchange <b>330</b> has an analog telephone <b>310</b> and a digital telephone <b>320</b> which are connected to it, and it provides exchange connections between one extension telephone <b>310</b> and another extension telephone <b>320</b> or between the extension telephones <b>310</b> and <b>320</b> and external telephone lines (telephone lines of telephone office via).

p-0072Here, the analog telephone <b>310</b> is a telephone used in a general home and is connected to the key telephone/private exchange <b>330</b> through an analog interface. The digital telephone <b>320</b> is a special telephone which is manufactured in order to make full use of the key telephone/private exchange, and is connected to key telephone/private exchange <b>330</b> through a digital interface.

p-0073The key telephone/private exchange <b>330</b> receives, from a manager, information on whether the connected telephone is the analog telephone <b>310</b> or the digital telephone <b>320</b>, makes a database by correlating the information with the extension numbers and stores the database in it. When a call setup request is made from the connected telephones <b>310</b> and <b>320</b>, the key telephone/private exchange <b>330</b> looks up the extension number and determines whether the telephone currently requesting the call setup is the analog telephone <b>310</b> or the digital telephone <b>320</b>.

p-0074The information on the kind of telephone correlated with the extension number is transmitted to the gatekeeper <b>350</b> coupled tightly with the key telephone/private exchange <b>330</b> and is managed in it.

p-0075The key telephone/private exchange <b>330</b> manages a gain table shown in Table 1, below, and this gain table is used to determine an amplification ratio when the PCM voice data are amplified on the VoIP gateway <b>340</b> in accordance with the kind of sender telephones <b>310</b> and <b>320</b>.

p-0076When the key telephone/private exchange <b>330</b> senses a call setup request from an extension line, it determines whether the sender telephone is an analog telephone <b>310</b> or a digital telephone <b>320</b> by identifying the extension telephone number. And, the key telephone/private exchange <b>330</b> looks up an IP address translation table transmitted from the gatekeeper <b>350</b> so as to be a database and managed by it, and determines whether the receiver (called) telephone <b>360</b> is an IP telephone <b>360</b> or a VoIP gateway <b>400</b>. Here, using the called telephone number, the key telephone/private exchange finds the IP address corresponding to it and determines whether the receiver is an IP telephone <b>360</b> or a VoIP gateway <b>400</b>. On the basis of the determination result, the key telephone/private exchange <b>330</b> looks up the gain table of Table 1, identifies the gain value and transmits the identified gain value to the VoIP gateway <b>340</b>.

p-0077The key telephone/private exchange <b>330</b> transmits the gain table information of Table 1 to the gatekeeper <b>350</b> so as to be stored in a database and managed by it.

p-0078<tables id="TABLE-US-00001" num="00001"><table frame="none" colsep="0" rowsep="0"><tgroup align="left" colsep="0" rowsep="0" cols="5"><colspec colname="1" colwidth="49pt" align="left" /><colspec colname="2" colwidth="42pt" align="left" /><colspec colname="3" colwidth="35pt" align="left" /><colspec colname="4" colwidth="42pt" align="left" /><colspec colname="5" colwidth="49pt" align="left" /><thead><row><entry namest="1" nameend="5" rowsep="1">TABLE 1</entry></row><row><entry namest="1" nameend="5" align="center" rowsep="1" /></row><row><entry>Receiver/</entry><entry>Analog</entry><entry>Digital</entry><entry /><entry>External VoIP</entry></row><row><entry>sender</entry><entry>telephone</entry><entry>telephone</entry><entry>IP telephone</entry><entry>gateway</entry></row><row><entry namest="1" nameend="5" align="center" rowsep="1" /></row></thead><tbody valign="top"><row><entry>Analog</entry><entry>X</entry><entry>X</entry><entry>9 db</entry><entry>10 db</entry></row><row><entry>telephone</entry></row><row><entry>Digital</entry><entry>X</entry><entry>X</entry><entry>8 db</entry><entry>13 db</entry></row><row><entry>telephone</entry></row><row><entry>IP telephone</entry><entry> 9 db</entry><entry> 8 db</entry><entry>X</entry><entry>X</entry></row><row><entry>External VoIP</entry><entry>10 db</entry><entry>13 db</entry><entry>X</entry><entry>X</entry></row><row><entry>gateway</entry></row><row><entry namest="1" nameend="5" align="center" rowsep="1" /></row></tbody></tgroup></table></tables>

p-0079As described above, the key telephone/private exchange <b>330</b> is installed in a building or a predetermined place and employs an external telephone line ‘endowed a telephone number’ from the telephone station. And, the key telephone/private exchange <b>330</b> enables the extension lines endowed their own numbers to communicate freely using the analog telephone <b>310</b> or the digital telephone <b>320</b> and provides various functions including switchover, holding, broadcasting, conference, and so on so that users can perform their businesses with ease and efficiency. The key telephone/private exchange <b>330</b> also identifies gain values and informs the VoIP gateway <b>340</b> of the gain values. So, the VoIP gateway <b>340</b> can look up the gain values when it compresses PCM voice data.

p-0080The key telephone/private exchange <b>330</b> can be connected to the VoIP gateway <b>340</b> using a digital line such as an E1/T1 <b>370</b> and an analog line such as a loop line.

p-0081The key telephone/private exchange <b>330</b> can be connected to another VoIP gateway <b>400</b> outside through a PSTN (public switch telephone network) <b>401</b> using an E1/T1 <b>390</b> or a loop line as an external telephone line trunk.

p-0082The VoIP gateway <b>340</b> is adapted to connect the PSTN to the VoIP network. Generally, in order to embody a VoIP voice communication, it is needed to employ protocols for controlling is call, for example, a protocol SS7 (Signaling System 7) for controlling call of the PSTN, an H.323 protocol for Internet, an SIP (Session Initiation Protocol) and so on.

p-0083The VoIP gateway <b>340</b> is needed to control an inter-transformation between a call control protocol used for an Internet and a PSTN when both networks are interworked and the media.

p-0084Generally, the VoIP gateway <b>340</b> can be classified according to its service. For example, the VoIP gateway <b>340</b> includes a built-in type gateway which is mounted on a Key Telephone System (KTS) or a Private Branch exchange (PBX) <b>330</b> as a card form, a server type gateway which is mounted on a platform such as a window Network Terminal (NT), a stand-alone type gateway which is independently constructed from other terminals, and so on.

p-0085The stand-alone type gateway can be classified into a TANDEM (Trunk and ENM (ear &amp; mouth)) function and a stand-alone function. The TANDEM function stand-alone gateway is a stand-alone gateway for supporting an interworking between heterogeneous lines.

p-0086The TANDEM function stand-alone gateway is connected to a private exchange and/or a key telephone system <b>330</b> through an internal T1/E1 interface, a loop start trunk interface and an SLC (Subscriber Line Circuit) interface.

p-0087The stand-alone type gateway of stand-alone function is connected to a plurality of telephone terminals directly. Accordingly, in connection with the present invention, the VoIP gateway <b>340</b> may be the built-in type gateway and the server type gateway which is mounted on a platform such as a window NT (window Network Terminal), and a TANDEM type gateway among the stand-alone type gateways which are independently constructed from other terminals.

p-0088The VoIP gateway <b>340</b> is connected to gatekeeper <b>350</b> through an Internet (IP) network <b>380</b>.

p-0089Main functions of the VoIP gateway <b>340</b> are to compress the PCM voice data received from the key telephone/private exchange <b>330</b>, transform the data into VoIP packets and transmit them to over the Internet network <b>380</b>, or transform the VoIP packet received from the Internet network <b>380</b> into PCM data and transfer the data to the key telephone/private exchange <b>330</b>.

p-0090Here, the VoIP gateway <b>340</b> stores a gain value transmitted from the key telephone/private exchange <b>330</b>. And then, in case that a call setup is completed and there exists a voice data exchange, when transforming PCM voice data into a VoIP packet, the VoIP gateway <b>340</b> amplifies and transforms the PCM voice data according to the stored gain value. And, in case of transforming the VoIP packet into the PCM voice data and outputting them, the key telephone/private exchange <b>330</b> amplifies the transformed PCM voice data and outputs them.

p-0091Of course, in case that a sender is not the key telephone/private exchange <b>330</b> but an IP telephone <b>360</b> or an external VoIP gateway <b>400</b>, the VoIP gateway <b>340</b> stores a gain value transmitted from the gatekeeper <b>350</b>, and then in case that the call setup is completed and there exists the voice data exchange, when transforming the PCM voice data into the VoIP packet, amplifies and outputs the PCM data according to the stored gain value. And, in case of transforming the VoIP packet into the PCM voice data and outputting the data, the VoIP gateway <b>340</b> amplifies the transformed PCM voice data and outputs them.

p-0092The gatekeeper <b>350</b> is one of H.323 Entity which is defined in the H.323 protocol being a multimedia communication standard of ITU-T, which is an apparatus for controlling, managing and integrating H.323 end points (gateway, terminal, MCU, and so on) existing in a packet-based network after making them one control area defined as a Zone.

p-0093Main functions of the gatekeeper <b>350</b> include an address translation function for translating the Alias name or a destination name into a network (IP) address name, a bandwidth control function of a call authentication (RAS) function for properly distributing a protocol related with the gatekeeper <b>350</b> and a bandwidth being a limited resource of a network to each end point in the H.323 of Registration/Admission/Status (RAS) and checking if they reaches to a limit values and then performing a blocking, a call control function for connecting/releasing call between one end point and another end point, and additional maintenance functions such as billing, statistics, and so on.

p-0094Such a gatekeeper <b>350</b> is connected to the IP telephone <b>360</b> through the VoIP gateways <b>340</b> and <b>400</b> through the IP network <b>380</b>.

p-0095The gatekeeper <b>350</b> manages the IP address translation table used for mapping the Internet telephone number and the IP address in order to perform the address translation function, which transmits IP address translation table information to the key telephone/private exchange <b>330</b> so that the key telephone/private exchange <b>330</b> can make the IP address translation table its database to be stored and managed.

p-0096When the gatekeeper <b>350</b> receives a call setup request for sharing the VoIP gateway <b>340</b> from the IP telephone <b>360</b> or the external VoIP gateway <b>400</b>, it analyses a sender IP address and determines whether the sender is the IP telephone <b>360</b> or the external VoIP gateway <b>400</b>. The gatekeeper <b>350</b> identifies information on the type of telephone related the extension number which is transmitted from the key telephone/private exchange <b>330</b> (here, called extension telephone number is used) and determines whether the called extension telephone is the analog telephone <b>310</b> or the digital telephone <b>320</b>.

p-0097Then, the gatekeeper <b>350</b> obtains the gain value by looking up the gain table (Table 1) transmitted from the key telephone/private exchange <b>330</b>, and transmits the obtained gain value to the VoIP gateway <b>340</b>.

p-0098Here, the IP telephone <b>360</b> is also called an Internet telephone which enables users to perform a voice communication in the VoIP net.

p-0099Now, an operation of the VoIP system in accordance with an embodiment of the present invention having the construction described above will be explained.

p-0100When a user makes a phone call using the analog telephone <b>310</b> or the digital telephone <b>320</b> and a call passing the VoIP gateway <b>340</b> is generated, the key telephone/private exchange <b>330</b> identifies the extension number of the telephone making a phone call and determines if the sender telephone is the analog telephone <b>310</b> or the digital telephone <b>320</b>.

p-0101Then, the key telephone/private exchange <b>330</b> searches for a called telephone number by looking up the IP address translation table transmitted from the gatekeeper <b>350</b> and determines if the receiver is the IP telephone <b>360</b> or the external VoIP gateway <b>400</b>.

p-0102After then, the key telephone/private exchange <b>330</b> identifies a gain value by looking up the gain table (Table 1) and transmits the gain value together with a call setup signal. Of course, the key telephone/private exchange <b>330</b> may transmit the call setup signal to the VoIP gateway <b>340</b> first, and then transmit the gain value using a special message.

p-0103As an example, when a user makes a phone call to the IP telephone <b>360</b> using the analog telephone <b>310</b>, the key telephone/private exchange <b>330</b> transmits a gain value of 9 db to the VoIP gateway <b>340</b>.

p-0104As an example, also, when a user makes a phone call to the external VoIP gateway <b>400</b> using the digital telephone <b>320</b>, the key telephone/private exchange <b>330</b> transmits a gain value of 13 db to the VoIP gateway <b>340</b>.

p-0105The VoIP gateway <b>340</b> which has received a call setup signal including a gain value from the key telephone/private exchange <b>330</b> extracts the gain value included in the call setup signal transmitted and stores it in a memory, and transmits the call setup message to the IP telephone <b>360</b> or the external VoIP gateway <b>400</b>. Of course, in case of receiving the gain value from the key telephone/private exchange <b>330</b> through a special message, it is possible to extract the gain value from the message transmitted and store it in a memory.

p-0106When the VoIP gateway <b>340</b> tries to compress and transmit the PCM voice data using a codec in case that a call setup is normally made to the IP telephone <b>360</b> or the external VoIP gateway <b>400</b>, it amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value before compressing them and then compresses and transmits the amplified PCM data using the codec.

p-0107Also, the VoIP gateway <b>340</b> transforms the VoIP packet transmitted from the called IP telephone <b>360</b> or external VoIP gateway <b>400</b> into the PCM voice data and then, according to the amplification ratio determined in response to the stored gain value, amplifies and transmits the PCM voice data to the key telephone/private exchange <b>330</b>.

p-0108In case that an external call is received, for example, when the VoIP gateway <b>340</b> receives a call setup signal transmitted from the IP telephone <b>360</b> or the VoIP gateway <b>400</b>, the call setup signal passes through the gatekeeper <b>350</b>. Here, the gatekeeper <b>350</b> identifies the gain value and transmits it to the VoIP gateway <b>340</b>.

p-0109At first, when the gatekeeper <b>350</b> receives a signal for requesting a call setup with the telephones <b>310</b> and <b>320</b> connected to the key telephone/private exchange <b>330</b> from the IP telephone <b>360</b> or the external VoIP gateway <b>400</b>, it identifies a sender IP address and determines whether the sender is the IP telephone <b>360</b> or the VoIP gateway <b>400</b>.

p-0110The gatekeeper <b>350</b> identifies an IP address included in the call setup signal and determines whether the sender requesting the call setup is the IP telephone <b>360</b> or the VoIP gateway <b>400</b>.

p-0111The gatekeeper <b>350</b> identifies the called telephone number and determines whether the receiver is the analog telephone <b>310</b> or the digital telephone <b>320</b> using information on the type of telephone related with the extension telephone number received from the key telephone/private exchange <b>330</b>.

p-0112The gatekeeper <b>350</b> obtains a gain value by looking up the gain table (Table 1) transmitted from the key telephone/private exchange <b>330</b>, and transmits the obtained gain value to the VoIP gateway <b>340</b>.

p-0113Then, in case that a call setup is normally made to the analog telephone <b>310</b> or the digital telephone <b>320</b>, the VoIP gateway <b>340</b> transforms the VoIP packet into PCM voice data and then, according to the gain value, amplifies and outputs the transformed PCM voice data.

p-0114Also, when the VoIP gateway <b>340</b> tries to compress and transmit the PCM voice data using the codec, it amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value before compressing the PCM voice data, and then compresses and transmits the amplified PCM data using the codec.

p-0115On the other hand, though the gain value of this embodiment is determined in consideration of the sender and the receiver both, it may be possible to determine the gain value by merely referring to the type of telephones <b>310</b> and <b>320</b> connected to the key telephone/private exchange <b>330</b>.

p-0116<figref idrefs="DRAWINGS">FIG. 4</figref> is a view showing an inner block diagram of a VoIP gateway.

p-0117Referring to <figref idrefs="DRAWINGS">FIG. 4</figref>, the VoIP gateway includes a subscriber line connector <b>402</b>, a switch <b>403</b> for connecting between subscribers who are connected through lines, a media processor <b>405</b> for compressing and decompressing common voice, a PSTN connector <b>407</b> for connecting to the PSTN and receiving an E1/T1 digital line of a key telephone/private exchange <b>408</b>, an IP network connector <b>406</b> for connecting to the IP network <b>409</b>, and a main controller <b>404</b>.

p-0118In the VoIP gateway described above, when the PSTN connector <b>407</b> interfaced with the key telephone/private exchange <b>408</b> requires the main controller <b>404</b> to make a call setup with the IP network <b>409</b>, the main controller <b>404</b> checks the state of the IP network <b>409</b> and then performs the call setup to the IP network <b>409</b> through the IP network connector <b>406</b>.

p-0119The media processor <b>405</b> compresses the PCM voice data inputted from the key telephone/private exchange <b>408</b> through the PSTN connector <b>407</b>, and then transmits the data to the IP network connector <b>406</b>. And, the media processor <b>405</b> also decompresses packet data inputted from the IP connector <b>406</b> and relays the call to the key telephone/private exchange <b>408</b> through the PSTN connector <b>407</b>.

p-0120Here, the main controller <b>404</b> extracts and stores a gain value included in a call setup message received from the key telephone/private exchange <b>408</b> (a special message may be used) and then provides the media processor <b>405</b> with the stored gain value when the call setup is completed.

p-0121Then, the media processor <b>405</b> amplifies the PCM voice data inputted according to the gain value, and compresses and transmits the amplified PCM voice data as VoIP packets.

p-0122Also, the media processor <b>405</b> transforms the VoIP packet into the PCM voice data according to the gain value, and then amplifies and outputs the transformed PCM voice data.

p-0123On the other hand, when the main controller <b>404</b> received an external call setup signal through the IP network connector <b>406</b> (of course, the main controller may receives the gain value from the gatekeeper using a special message), it extracts the gain value included in the call setup signal and stores the value, and provides the media processor <b>405</b> with the gain value stored after the call setup is completed.

p-0124Then, the media processor <b>405</b> transforms the VoIP packet into the PCM voice data according to the gain value, and then amplifies and outputs the transformed PCM voice data.

p-0125Also, the media processor <b>405</b> amplifies the PCM voice data inputted according to the gain value, and then compresses and transmits the amplified PCM voice data as VoIP packets.

p-0126<figref idrefs="DRAWINGS">FIG. 5</figref> is a view showing a detailed block construction of the IP network connector and the media processor shown in <figref idrefs="DRAWINGS">FIG. 4</figref> and connected to PSTN connector <b>407</b> and key telephone/private exchange <b>408</b>.

p-0127Referring to <figref idrefs="DRAWINGS">FIG. 5</figref>, the IP network connector <b>406</b> includes a central processing unit (CPU) <b>511</b>, a memory <b>512</b> and a Compact Peripheral Component Interconnect (cPCI) bridge <b>513</b>. And, the media processor <b>405</b> includes a digital signal processor (DSP) <b>521</b>, a central processing unit (CPU) <b>522</b>, a memory <b>523</b>, and a cPCI bridge <b>524</b>. The cPCI bridges <b>513</b> and <b>524</b> are constituents used to match CPU busses.

p-0128Here, since a normal media processor <b>405</b> has a lower degree of integration of a channel than the IP network connector <b>406</b>, it is common that one IP network connector <b>406</b> is matched with a number of media processors <b>405</b>. At that time, there occurs a task that a number of hardware PBA (Printed Board Assembly) should be matched with a common bus for voice traffic in a backplane in order that the IP network connector <b>406</b> transmits and receives the voice traffic to and from the media processor <b>405</b>.

p-0129It is common to use a cPCI bus in the aspect of a bus band and operating with a current level of technology, and it is necessary to use the CPUs <b>511</b> and <b>522</b> for operating the bus in case of using the cPCI bus. Here, the CPUs <b>511</b> and <b>522</b> should be used for transferring voice traffic and also for operating the bus.

p-0130The CPU <b>511</b> of the IP network connector <b>406</b> is used to process the IP protocol and to operate the cPCI bus, and the cPCI bridges <b>513</b> is used to match the CPU bus with cPCI bus.

p-0131The DSP <b>521</b> of the media processor <b>405</b> is an essential constituent for embodying vocoding function, and the CPU <b>522</b> of the media processor <b>405</b> is an essential constituent for controlling the cPCI bridge <b>524</b>.

p-0132The memories <b>512</b> and <b>523</b> are essential constituents which act as buffers for transmitting and receiving data processed in the CPUs <b>511</b> and <b>522</b> and the DSP <b>521</b>.

p-0133The cPCI bus is made up of a master and a number of targets wherein an IP network connector <b>406</b> acts as the master and a number of media processors act as the targets, in the conventional art.

p-0134In connection with the present invention, the CPU <b>522</b> of the media processor <b>405</b> receives a gain value from the main controller <b>404</b> of the VoIP gateway and stores it in the memory <b>523</b>, and in case of compressing the PCM data inputted from a PSTN connector <b>407</b> into the VoIP packet, controls the DSP <b>521</b> so that the DSP <b>521</b> amplifies the PCM data according to the gain value and then compresses the amplified data.

p-0135When the CPU <b>522</b> of the media processor <b>405</b> transforms the VoIP packet inputted from the IP network connector <b>406</b> (<figref idrefs="DRAWINGS">FIG. 4</figref>) into the PCM data, it controls the DSP <b>521</b> so that the DSP <b>521</b> amplifies the PCM data according to the gain value stored and outputs the data.

p-0136<figref idrefs="DRAWINGS">FIG. 6</figref> is a view showing an operation flow chart of a method for providing a dynamic gain in a sender using a VoIP system in accordance with an embodiment of the present invention.

p-0137Referring to <figref idrefs="DRAWINGS">FIG. 6</figref>, when a telephone user makes a phone call which passes through a VoIP gateway using an analog telephone or a digital telephone, the analog telephone or the digital telephone transmit a call setup message including a called telephone number to a key telephone/private exchange (S<b>110</b>).

p-0138Then, the key telephone/private exchange identifies an extension number of a sender and determines whether the type of telephone of the sender is an analog telephone or a digital telephone.

p-0139Then, the key telephone/private exchange looks up an IP address translation table transmitted from a gatekeeper and determines whether the receiver is an IP telephone or an external VoIP gateway.

p-0140Then, according to the determination, the key telephone/private exchange obtains a gain value by looking up a gain table and then transmits the gain value together with the call setup message to the VoIP gateway (S<b>112</b>). The obtained gain value corresponds to either the extension telephone making the call, the terminal (end-point) receiving the call (i.e., the IP telephone or the external VoIP gateway) or both the extension telephone making the call and the terminal (end-point) receiving the call.

p-0141Then, the VoIP gateway extracts the gain value from the call setup message, stores the value (S<b>113</b>), and transmits the call setup message to the gatekeeper (S<b>114</b>). The call setup message is then transmitted to the called IP telephone or external VoIP gateway (S<b>116</b>).

p-0142Then, when the gatekeeper and the VoIP gateway receive a call response message from the IP telephone or the external VoIP gateway (S<b>118</b> and S<b>120</b>), the received call response message is transmitted to the key telephone/private exchange and the telephone (S<b>122</b> and S<b>124</b>).

p-0143When the VoIP gateway tries to compress and transmit the PCM voice data using the codec after the call setup has been completed and the call setup has been normally made to the IP telephone or the external VoIP gateway, the VoIP gateway amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value and then compresses and transmits the amplified PCM data using the codec.

p-0144Also, the VoIP gateway transforms the VoIP packet received from the IP telephone or the external VoIP gateway into the PCM voice data, amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value and transmits the amplified PCM voice data to the key telephone/private exchange.

p-0145<figref idrefs="DRAWINGS">FIG. 7</figref> is a view showing an operation flow chart of a method for providing a dynamic gain in a receiver using a VoIP system in accordance with another embodiment of the present invention.

p-0146Referring to <figref idrefs="DRAWINGS">FIG. 7</figref>, when a gatekeeper receives a call setup message form an IP telephone or an external VoIP gateway (S<b>210</b>), it identifies an IP address of a sender and determines whether the sender is an IP telephone or an external VoIP gateway by looking up an IP address translation table.

p-0147Then, the gatekeeper determines whether a receiver is an analog telephone or a digital telephone using information on the kind of telephone for an extension telephone number transmitted from the key telephone/private exchange. And then, the gatekeeper obtains a gain value by looking up the gain table and transmits the gain value obtained together with the call setup message to the VoIP gateway (S<b>212</b>).

p-0148Then, the VoIP gateway extracts the gain value from the call setup message, stores the gain value (S<b>213</b>) and transmits the call setup message to the key telephone/private exchange (S<b>214</b>). The call setup message is then transmitted to the analog telephone or the digital telephone (S<b>216</b>).

p-0149Then, when the VoIP gateway receives a call response message from the analog telephone or the digital telephone (S<b>218</b> and S<b>220</b>), the received call response message is transmitted to the IP telephone or the external VoIP gateway so that the call setup is made (S<b>222</b> and S<b>224</b>).

p-0150Then, as the call setup is completed, the VoIP gateway transforms the VoIP packet transmitted from the IP telephone or the external VoIP gateway into the PCM voice data, amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value, and then transmits the data to the key telephone/private exchange.

p-0151Also, when the VoIP gateway tries to compress the PCM voice data using the codec and transmit the PCM voice data to the IP telephone or the external VoIP gateway, it amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value, compresses the amplified PCM data using the codec, and then transmits them as VoIP packets.

p-0152Even though the present invention explains the case that the key telephone/private exchange, the VoIP gateway and the gatekeeper are close coupled and share information in the database, the same method will be applied to the case that the constituents are not coupled closely.

p-0153Although the preferred embodiments of the present invention have been disclosed for illustrative purposes, those skilled in the art appreciate that various modifications, additions and substitutions are possible, without departing from the scope and spirit of the invention as disclosed in the accompanying claims.

EFFECT

p-0154In accordance with the present invention, when connecting a call to communication equipment (an IP phone or a VoIP gateway) connected to an external IP network through another VoIP gateway in a key telephone/private exchange, a speech quality can be enhanced by enabling the call to have proper gain values according to the type of terminals to be connected to the call.